+#ifdef HAVE_GSTREAMER
+
+ /* note: this requires gstreamer 0.10.x and a big list of plugins. */
+ /* it's currently hardcoded to use a big-endian alsasink as sink. */
#include <lib/base/eerror.h>
#include <lib/base/object.h>
#include <lib/base/ebase.h>
#include <lib/service/service.h>
#include <lib/base/init_num.h>
#include <lib/base/init.h>
+#include <gst/gst.h>
// eServiceFactoryMP3
{
ePtr<eServiceCenter> sc;
- eServiceCenter::getInstance(sc);
+ eServiceCenter::getPrivInstance(sc);
if (sc)
sc->addServiceFactory(eServiceFactoryMP3::id, this);
{
ePtr<eServiceCenter> sc;
- eServiceCenter::getInstance(sc);
+ eServiceCenter::getPrivInstance(sc);
if (sc)
sc->removeServiceFactory(eServiceFactoryMP3::id);
}
return 0;
}
+RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &, ePtr<iServiceOfflineOperations> &ptr)
+{
+ ptr = 0;
+ return -1;
+}
+
+
// eStaticServiceMP3Info
RESULT eStaticServiceMP3Info::getName(const eServiceReference &ref, std::string &name)
{
- name = "MP3 file: " + ref.path;
+ size_t last = ref.path.rfind('/');
+ if (last != std::string::npos)
+ name = ref.path.substr(last+1);
+ else
+ name = ref.path;
return 0;
}
-// eServiceMP3
-
-void eServiceMP3::test_end()
+int eStaticServiceMP3Info::getLength(const eServiceReference &ref)
{
- eDebug("end of mp3!");
- stop();
+ return -1;
}
-eServiceMP3::eServiceMP3(const char *filename): filename(filename), test(eApp)
+// eServiceMP3
+
+eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
{
+ m_stream_tags = 0;
+ CONNECT(m_pump.recv_msg, eServiceMP3::gstPoll);
+ GstElement *source = 0;
+
+ GstElement *filter = 0, *decoder = 0, *conv = 0, *flt = 0, *sink = 0; /* for audio */
+
+ GstElement *audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *mpegdemux = 0;
+
m_state = stIdle;
eDebug("SERVICEMP3 construct!");
+
+ /* FIXME: currently, decodebin isn't possible for
+ video streams. in that case, make a manual pipeline. */
+
+ const char *ext = strrchr(filename, '.');
+ if (!ext)
+ ext = filename;
+
+ int is_mpeg_ps = !(strcasecmp(ext, ".mpeg") && strcasecmp(ext, ".mpg") && strcasecmp(ext, ".vob") && strcasecmp(ext, ".bin"));
+ int is_mpeg_ts = !strcasecmp(ext, ".ts");
+ int is_mp3 = !strcasecmp(ext, ".mp3"); /* force mp3 instead of decodebin */
+ int is_video = is_mpeg_ps || is_mpeg_ts;
+ int is_streaming = !strncmp(filename, "http://", 7);
+
+ eDebug("filename: %s, is_mpeg_ps: %d, is_mpeg_ts: %d, is_video: %d, is_streaming: %d, is_mp3: %d", filename, is_mpeg_ps, is_mpeg_ts, is_video, is_streaming, is_mp3);
+
+ int is_audio = !is_video;
+
+ int all_ok = 0;
+
+ m_gst_pipeline = gst_pipeline_new ("audio-player");
+ if (!m_gst_pipeline)
+ eWarning("failed to create pipeline");
+
+ if (!is_streaming)
+ source = gst_element_factory_make ("filesrc", "file-source");
+ else
+ {
+ source = gst_element_factory_make ("neonhttpsrc", "http-source");
+ if (source)
+ g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
+ }
+
+ if (!source)
+ eWarning("failed to create %s", is_streaming ? "neonhttpsrc" : "filesrc");
+ else
+ /* configure source */
+ g_object_set (G_OBJECT (source), "location", filename, NULL);
+
+ if (is_audio)
+ {
+ /* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
+ const char *decodertype = is_mp3 ? "mad" : "decodebin";
+
+ decoder = gst_element_factory_make (decodertype, "decoder");
+ if (!decoder)
+ eWarning("failed to create %s decoder", decodertype);
+
+ /* mp3 decoding needs id3demux to extract ID3 data. 'decodebin' would do that internally. */
+ if (is_mp3)
+ {
+ filter = gst_element_factory_make ("id3demux", "filter");
+ if (!filter)
+ eWarning("failed to create id3demux");
+ }
+
+ conv = gst_element_factory_make ("audioconvert", "converter");
+ if (!conv)
+ eWarning("failed to create audioconvert");
+
+ flt = gst_element_factory_make ("capsfilter", "flt");
+ if (!flt)
+ eWarning("failed to create capsfilter");
+
+ /* for some reasons, we need to set the sample format to depth/width=16, because auto negotiation doesn't work. */
+ /* endianness, however, is not required to be set anymore. */
+ if (flt)
+ {
+ GstCaps *caps = gst_caps_new_simple("audio/x-raw-int", /* "endianness", G_TYPE_INT, 4321, */ "depth", G_TYPE_INT, 16, "width", G_TYPE_INT, 16, "channels", G_TYPE_INT, 2, (char*)0);
+ g_object_set (G_OBJECT (flt), "caps", caps, (char*)0);
+ gst_caps_unref(caps);
+ }
+
+ sink = gst_element_factory_make ("alsasink", "alsa-output");
+ if (!sink)
+ eWarning("failed to create osssink");
+
+ if (source && decoder && conv && sink)
+ all_ok = 1;
+ } else /* is_video */
+ {
+ /* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
+ | queue_video -> dvbvideosink */
+
+ audio = gst_element_factory_make("dvbaudiosink", "audio");
+ queue_audio = gst_element_factory_make("queue", "queue_audio");
+
+ video = gst_element_factory_make("dvbvideosink", "video");
+ queue_video = gst_element_factory_make("queue", "queue_video");
+
+ if (is_mpeg_ps)
+ mpegdemux = gst_element_factory_make("flupsdemux", "mpegdemux");
+ else
+ mpegdemux = gst_element_factory_make("flutsdemux", "mpegdemux");
+
+ if (!mpegdemux)
+ {
+ eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");
+ mpegdemux = gst_element_factory_make("mpegdemux", "mpegdemux");
+ }
+
+ eDebug("audio: %p, queue_audio %p, video %p, queue_video %p, mpegdemux %p", audio, queue_audio, video, queue_video, mpegdemux);
+ if (audio && queue_audio && video && queue_video && mpegdemux)
+ {
+ g_object_set (G_OBJECT (queue_audio), "max-size-buffers", 0, NULL);
+ g_object_set (G_OBJECT (queue_audio), "max-size-time", (guint64)0, NULL);
+ g_object_set (G_OBJECT (queue_video), "max-size-buffers", 0, NULL);
+ g_object_set (G_OBJECT (queue_video), "max-size-time", (guint64)0, NULL);
+ all_ok = 1;
+ }
+ }
+
+ if (m_gst_pipeline && all_ok)
+ {
+ gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline)), gstBusSyncHandler, this);
+
+ if (is_audio)
+ {
+ if (!is_mp3)
+ {
+ /* decodebin has dynamic pads. When they get created, we connect them to the audio bin */
+ g_signal_connect (decoder, "new-decoded-pad", G_CALLBACK(gstCBnewPad), this);
+ g_signal_connect (decoder, "unknown-type", G_CALLBACK(gstCBunknownType), this);
+ }
+
+ /* gst_bin will take the 'floating references' */
+ gst_bin_add_many (GST_BIN (m_gst_pipeline),
+ source, decoder, NULL);
+
+ if (filter)
+ {
+ /* id3demux also has dynamic pads, which need to be connected to the decoder (this is done in the 'gstCBfilterPadAdded' CB) */
+ gst_bin_add(GST_BIN(m_gst_pipeline), filter);
+ gst_element_link(source, filter);
+ m_decoder = decoder;
+ g_signal_connect (filter, "pad-added", G_CALLBACK(gstCBfilterPadAdded), this);
+ } else
+ /* in decodebin's case we can just connect the source with the decodebin, and decodebin will take care about id3demux (or whatever is required) */
+ gst_element_link(source, decoder);
+
+ /* create audio bin with the audioconverter, the capsfilter and the audiosink */
+ m_gst_audio = gst_bin_new ("audiobin");
+
+ GstPad *audiopad = gst_element_get_pad (conv, "sink");
+ gst_bin_add_many(GST_BIN(m_gst_audio), conv, flt, sink, (char*)0);
+ gst_element_link_many(conv, flt, sink, (char*)0);
+ gst_element_add_pad(m_gst_audio, gst_ghost_pad_new ("sink", audiopad));
+ gst_object_unref(audiopad);
+ gst_bin_add (GST_BIN(m_gst_pipeline), m_gst_audio);
+
+ /* in mad's case, we can directly connect the decoder to the audiobin. otherwise, we do this in gstCBnewPad */
+ if (is_mp3)
+ gst_element_link(decoder, m_gst_audio);
+ } else
+ {
+ gst_bin_add_many(GST_BIN(m_gst_pipeline), source, mpegdemux, audio, queue_audio, video, queue_video, NULL);
+ gst_element_link(source, mpegdemux);
+ gst_element_link(queue_audio, audio);
+ gst_element_link(queue_video, video);
+
+ m_gst_audioqueue = queue_audio;
+ m_gst_videoqueue = queue_video;
+
+ g_signal_connect(mpegdemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+ }
+ } else
+ {
+ if (m_gst_pipeline)
+ gst_object_unref(GST_OBJECT(m_gst_pipeline));
+ if (source)
+ gst_object_unref(GST_OBJECT(source));
+ if (decoder)
+ gst_object_unref(GST_OBJECT(decoder));
+ if (conv)
+ gst_object_unref(GST_OBJECT(conv));
+ if (sink)
+ gst_object_unref(GST_OBJECT(sink));
+
+ if (audio)
+ gst_object_unref(GST_OBJECT(audio));
+ if (queue_audio)
+ gst_object_unref(GST_OBJECT(queue_audio));
+ if (video)
+ gst_object_unref(GST_OBJECT(video));
+ if (queue_video)
+ gst_object_unref(GST_OBJECT(queue_video));
+ if (mpegdemux)
+ gst_object_unref(GST_OBJECT(mpegdemux));
+
+ eDebug("sorry, can't play.");
+ m_gst_pipeline = 0;
+ }
+
+ gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
}
eServiceMP3::~eServiceMP3()
{
- eDebug("SERVICEMP3 destruct!");
if (m_state == stRunning)
stop();
+
+ if (m_stream_tags)
+ gst_tag_list_free(m_stream_tags);
+
+ if (m_gst_pipeline)
+ {
+ gst_object_unref (GST_OBJECT (m_gst_pipeline));
+ eDebug("SERVICEMP3 destruct!");
+ }
}
DEFINE_REF(eServiceMP3);
assert(m_state == stIdle);
m_state = stRunning;
-
- printf("mp3 starts\n");
- printf("MP3: %s start\n", filename.c_str());
- test.start(1000, 1);
- CONNECT(test.timeout, eServiceMP3::test_end);
+ if (m_gst_pipeline)
+ {
+ eDebug("starting pipeline");
+ gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
+ }
m_event(this, evStart);
return 0;
}
assert(m_state != stIdle);
if (m_state == stStopped)
return -1;
- test.stop();
- printf("MP3: %s stop\n", filename.c_str());
+ printf("MP3: %s stop\n", m_filename.c_str());
+ gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
m_state = stStopped;
- m_event(this, evEnd);
return 0;
}
-RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr) { ptr=this; return 0; }
+RESULT eServiceMP3::setTarget(int target)
+{
+ return -1;
+}
+
+RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr)
+{
+ ptr=this;
+ return 0;
+}
+RESULT eServiceMP3::setSlowMotion(int ratio)
+{
+ return -1;
+}
+
+RESULT eServiceMP3::setFastForward(int ratio)
+{
+ return -1;
+}
+
// iPausableService
-RESULT eServiceMP3::pause() { printf("mp3 pauses!\n"); return 0; }
-RESULT eServiceMP3::unpause() { printf("mp3 unpauses!\n"); return 0; }
+RESULT eServiceMP3::pause()
+{
+ if (!m_gst_pipeline)
+ return -1;
+ gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED);
+ return 0;
+}
+
+RESULT eServiceMP3::unpause()
+{
+ if (!m_gst_pipeline)
+ return -1;
+ gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
+ return 0;
+}
+
+ /* iSeekableService */
+RESULT eServiceMP3::seek(ePtr<iSeekableService> &ptr)
+{
+ ptr = this;
+ return 0;
+}
-RESULT eServiceMP3::info(ePtr<iServiceInformation>&i) { i = this; return 0; }
+RESULT eServiceMP3::getLength(pts_t &pts)
+{
+ if (!m_gst_pipeline)
+ return -1;
+ if (m_state != stRunning)
+ return -1;
+
+ GstFormat fmt = GST_FORMAT_TIME;
+ gint64 len;
+
+ if (!gst_element_query_duration(m_gst_pipeline, &fmt, &len))
+ return -1;
+
+ /* len is in nanoseconds. we have 90 000 pts per second. */
+
+ pts = len / 11111;
+ return 0;
+}
+
+RESULT eServiceMP3::seekTo(pts_t to)
+{
+ if (!m_gst_pipeline)
+ return -1;
+
+ /* convert pts to nanoseconds */
+ gint64 time_nanoseconds = to * 11111LL;
+ if (!gst_element_seek (m_gst_pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
+ GST_SEEK_TYPE_SET, time_nanoseconds,
+ GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE))
+ {
+ eDebug("SEEK failed");
+ return -1;
+ }
+ return 0;
+}
+
+RESULT eServiceMP3::seekRelative(int direction, pts_t to)
+{
+ if (!m_gst_pipeline)
+ return -1;
+
+ pause();
+
+ pts_t ppos;
+ getPlayPosition(ppos);
+ ppos += to * direction;
+ if (ppos < 0)
+ ppos = 0;
+ seekTo(ppos);
+
+ unpause();
+
+ return 0;
+}
+
+RESULT eServiceMP3::getPlayPosition(pts_t &pts)
+{
+ if (!m_gst_pipeline)
+ return -1;
+ if (m_state != stRunning)
+ return -1;
+
+ GstFormat fmt = GST_FORMAT_TIME;
+ gint64 len;
+
+ if (!gst_element_query_position(m_gst_pipeline, &fmt, &len))
+ return -1;
+
+ /* len is in nanoseconds. we have 90 000 pts per second. */
+ pts = len / 11111;
+ return 0;
+}
+
+RESULT eServiceMP3::setTrickmode(int trick)
+{
+ /* trickmode currently doesn't make any sense for us. */
+ return -1;
+}
+
+RESULT eServiceMP3::isCurrentlySeekable()
+{
+ return 1;
+}
+
+RESULT eServiceMP3::info(ePtr<iServiceInformation>&i)
+{
+ i = this;
+ return 0;
+}
RESULT eServiceMP3::getName(std::string &name)
{
- name = "MP3 File: " + filename;
+ name = m_filename;
+ size_t n = name.rfind('/');
+ if (n != std::string::npos)
+ name = name.substr(n + 1);
return 0;
}
+int eServiceMP3::getInfo(int w)
+{
+ switch (w)
+ {
+ case sTitle:
+ case sArtist:
+ case sAlbum:
+ case sComment:
+ case sTracknumber:
+ case sGenre:
+ return resIsString;
+
+ default:
+ return resNA;
+ }
+}
+
+std::string eServiceMP3::getInfoString(int w)
+{
+ gchar *tag = 0;
+ switch (w)
+ {
+ case sTitle:
+ tag = GST_TAG_TITLE;
+ break;
+ case sArtist:
+ tag = GST_TAG_ARTIST;
+ break;
+ case sAlbum:
+ tag = GST_TAG_ALBUM;
+ break;
+ case sComment:
+ tag = GST_TAG_COMMENT;
+ break;
+ case sTracknumber:
+ tag = GST_TAG_TRACK_NUMBER;
+ break;
+ case sGenre:
+ tag = GST_TAG_GENRE;
+ break;
+ default:
+ return "";
+ }
+
+ if (!m_stream_tags || !tag)
+ return "";
+
+ gchar *value;
+
+ if (gst_tag_list_get_string(m_stream_tags, tag, &value))
+ {
+ std::string res = value;
+ g_free(value);
+ return res;
+ }
+
+ return "";
+}
+
+
+ void foreach(const GstTagList *list, const gchar *tag, gpointer user_data)
+ {
+ if (tag)
+ eDebug("Tag: %c%c%c%c", tag[0], tag[1], tag[2], tag[3]);
+
+ }
+
+void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
+{
+ if (msg)
+ {
+ gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
+ eDebug("gst_message: %s", string);
+ g_free(string);
+ }
+
+ switch (GST_MESSAGE_TYPE (msg))
+ {
+ case GST_MESSAGE_EOS:
+ m_event((iPlayableService*)this, evEOF);
+ break;
+ case GST_MESSAGE_ERROR:
+ {
+ gchar *debug;
+ GError *err;
+ gst_message_parse_error (msg, &err, &debug);
+ g_free (debug);
+ eWarning("Gstreamer error: %s", err->message);
+ g_error_free(err);
+ /* TODO: signal error condition to user */
+ break;
+ }
+ case GST_MESSAGE_TAG:
+ {
+ GstTagList *tags, *result;
+ gst_message_parse_tag(msg, &tags);
+
+ result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_PREPEND);
+ if (result)
+ {
+ if (m_stream_tags)
+ gst_tag_list_free(m_stream_tags);
+ m_stream_tags = result;
+ }
+ gst_tag_list_free(tags);
+
+ m_event((iPlayableService*)this, evUpdatedInfo);
+ break;
+ }
+ default:
+ break;
+ }
+}
+
+GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)
+{
+ eServiceMP3 *_this = (eServiceMP3*)user_data;
+ _this->m_pump.send(1);
+ /* wake */
+ return GST_BUS_PASS;
+}
+
+void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
+{
+ eServiceMP3 *_this = (eServiceMP3*)user_data;
+
+ gchar *name;
+ name = gst_pad_get_name (pad);
+ g_print ("A new pad %s was created\n", name);
+ if (!strncmp(name, "audio_", 6)) // mpegdemux uses video_nn with n=0,1,.., flupsdemux uses stream id
+ gst_pad_link(pad, gst_element_get_pad (_this->m_gst_audioqueue, "sink"));
+ if (!strncmp(name, "video_", 6))
+ gst_pad_link(pad, gst_element_get_pad (_this->m_gst_videoqueue, "sink"));
+ g_free (name);
+
+}
+
+void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
+{
+ eServiceMP3 *_this = (eServiceMP3*)user_data;
+ gst_pad_link(pad, gst_element_get_pad (_this->m_decoder, "sink"));
+}
+
+void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data)
+{
+ eServiceMP3 *_this = (eServiceMP3*)user_data;
+ GstCaps *caps;
+ GstStructure *str;
+ GstPad *audiopad;
+
+ /* only link once */
+ audiopad = gst_element_get_pad (_this->m_gst_audio, "sink");
+ if (GST_PAD_IS_LINKED (audiopad)) {
+ eDebug("audio already linked!");
+ g_object_unref (audiopad);
+ return;
+ }
+
+ /* check media type */
+ caps = gst_pad_get_caps (pad);
+ str = gst_caps_get_structure (caps, 0);
+ eDebug("gst new pad! %s", gst_structure_get_name (str));
+
+ if (!g_strrstr (gst_structure_get_name (str), "audio")) {
+ gst_caps_unref (caps);
+ gst_object_unref (audiopad);
+ return;
+ }
+
+ gst_caps_unref (caps);
+ gst_pad_link (pad, audiopad);
+}
+
+void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
+{
+ eServiceMP3 *_this = (eServiceMP3*)user_data;
+ GstStructure *str;
+
+ /* check media type */
+ caps = gst_pad_get_caps (pad);
+ str = gst_caps_get_structure (caps, 0);
+ eDebug("unknown type: %s - this can't be decoded.", gst_structure_get_name (str));
+ gst_caps_unref (caps);
+}
+
+void eServiceMP3::gstPoll(const int&)
+{
+ /* ok, we have a serious problem here. gstBusSyncHandler sends
+ us the wakup signal, but likely before it was posted.
+ the usleep, an EVIL HACK (DON'T DO THAT!!!) works around this.
+
+ I need to understand the API a bit more to make this work
+ proplerly. */
+ usleep(1);
+
+ GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline));
+ GstMessage *message;
+ while ((message = gst_bus_pop (bus)))
+ {
+ gstBusCall(bus, message);
+ gst_message_unref (message);
+ }
+}
+
eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
+#else
+#warning gstreamer not available, not building media player
+#endif